On Thu, 2009-06-11 at 11:49 +0700, Boy Aidil Sjam wrote: > > > -----Original Message----- > > From: Scott Lawrence [mailto:[email protected]] > > Sent: Wednesday, June 10, 2009 7:20 PM > > To: Boy Aidil Sjam > > Cc: [email protected] > > Subject: RE: [sipx-users] Interconnection 2 sipXecs > > > > On Wed, 2009-06-10 at 17:30 +0700, Boy Aidil Sjam wrote: > > > Scott, > > > I've setup lab environment before go live to public network > > (internet), and I like to see how it works. > > > There's no NAT involve in this lab testing. > > > The scenarios: > > > 1. I installed 2 sipXecs server, each server in their own domain. > > > 2. Each server use the same dial-plan (I set up ext. 200 - 201) > > > 3. I set up prefix 9 to dial to another domain. > > > 4. User from server A able to dial server B autoattendant using 9100 > > and able to transfer to another extension and vice versa. > > > 5. But when one of the user want to dial directly to user extension > > in another server (dialed 9200) the call directly terminated. > > > > > > Is that the way how the sipXecs works or is there something wrong > > with my configuration? > > > > That should work. > > > > > > To debug this, you'll need to trace the message flow and see > > what's going wrong. See: > > > > http://sipx- > > wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewer#Get > > ting_SIP_Messages_to_display > > > > when you get the trace data, take a look at it using sipviewer > > and/or post the trace with a description of your configuration > > (identify components by IP address), what you were doing, and > > which call in the trace you're talking about (by call-id or > > frame number in the trace, preferably). > > > > Server A > Domain: office.net > Hostname: voip.office.net > IP address: 10.10.12.50 > User Extension: 201 > User IP address: 10.10.12.169 > > Custom dial plan > Dialed Number --> Prefix 9 and 3 digits > Required Permission --> Local Dialing > Resulting Call --> Dial (empty) and append Matched suffix > Schedule Always > Gateway: voip.net.lab > > Server B > Domain: net.lab > Hostname: voip.net.lab > IP address: 10.10.12.200 > User extension: 201 > User IP address: 10.10.12.170 > > Custom dial plan > Dialed Number --> Prefix 9 and 3 digits > Required Permission --> Local Dialing > Resulting Call --> Dial (empty) and append Matched suffix > Schedule Always > Gateway: voip.office.net > > 1st test, extension [email protected] dialed 9100 ([email protected]) > Result: AA answered with default greeting > Then [email protected] press 201 > Result: Call transfer to [email protected], and the phone is ringing > > 2nd test, extension [email protected] dialed 9201 ([email protected]) > Result: Busy tone and followed by AA said, "The person you call isn't > available" > > I attached the 2nd test sipx-trace result from both server
It looks like you don't have the clocks synchronized on these two servers; this makes the traces very difficult to combine and analyze. >From what I can see, there was no phone registered at [email protected] _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
